Effect of the stimulation level on the refractory behavior of the electrically stimulated auditory nerve

The refractory behavior of the electrically stimulated auditory nerve can be described by the recovery function, which plots the ECAP amplitude in response to a masker/probe stimulus pair as a function of the time interval (Masker Probe Interval, MPI) between the two stimuli. The recovery function is characterized by two time intervals or periods: In the first interval (the Absolute Refractory Period, ARP), typically lasting for 300 to 400us, the neurons stimulated by the masker are in absolute refractory and unable to respond to the probe stimulus. As the MPI is gradually increased beyond the ARP, the stimulated neural population is increasingly able to respond to the probe stimulus (i.e. relative refractory) as the inhibitory effects of the masker diminishes. This second interval (the Relative Refractory Period, RRP) can be characterized by the time constant of an asymptotically increasing exponential function (Morsnowski et al. 2006). This recovery time constant provides an indication of the neurons’ temporal characteristics. Previous reports (e.g. Battmer et al. 2004) suggest that this time constant is affected by the stimulation level used to determine the recovery function. Such a dependency would make it difficult to characterize the refractory behavior of the stimulated neurons using the recovery function. In this study, the refractory behavior of the electrically stimulated auditory nerve with respect to stimulation level was examined retrospectively. It was expected that increasing the stimulation level would result in more deterministic behavior

Melody contour identification and instrument recognition using semitone mapping in Nucleus Cochlear Implant recipients

Cochlear implants (CIs) were originally aimed at restoring speech perception for patients with profound hearing loss. Many postlingually deafened CI patients report that music is not well perceived while others enjoy it. Music consists of complex sounds composed of tones with harmonic structure of overtones and temporal fine structure. The harmonic structure is not preserved by the current standard (Std) ACE (advanced combination encoders) mapping and the temporal fine structure is not well encoded. The mapping is believed to produce distortion due to compression oin the low frequency range. In 2008 we proposed two new semitone (Smt) mappings (Smt-LF and Smt-MF) in two frequency ranges (130-1502 Hz and 440-5040 Hz) respectively (Omran et al. 2008). Smt mapping is expected to preserve the harmonic structure representation of overtones and this may improve melody recognition with CIs. In this paper two psychoacoustic experiments (melody contour identification (MCI) (Galvin et al. 2007) and instrument recognition (IR)) were conducted with three different conditions (Std, Smt-MF and Smt-LF mappings) with CI recipients by streaming processed stimuli directly to the implant. The MCI test included five patterns (rising - rising falling - flat - falling rising – falling). Each pattern consisted of five tones; each tone had a fundamental frequency and four overtones. The lowest fundamental frequency of each pattern is called “root”. Signals had two different roots A3 (220 Hz) and A4 (440 Hz). Proposed nine patterns with three roots (A3, A4 and A5) by Galvin et al. (2007) were examined in a pilot test. This test took a long time and the preliminary results showed a possibility to reduce the number of patterns to five and eliminate the fifth octave root (A5). In the IR test, four pairs of instruments (Trumpet and Trombone, Flute and Clarinet, Violin and Cello, Guitar and Piano) from four groups (Brass, Woodwind, Struck and String instruments) respectively were used. MCI and IR tests were conducted with 8 CI recipients. Results from MCI tests showed an improvement with Smt mapping in respect to Std mapping or at least similar results. However, wrong identification occurred with notes having filtered out partials. CI recipients showed an increase in identifying melody contour patterns with Smt mappings. Instrument identification performance decreased with semitone mappings

TNRT profiles with the Nucleus Research Platform 8 system

This study investigates the effect of the Nucleus CI24RE implant's neural response telemetry (NRT) system, which has less internal noise compared to its predecessor, the CI24M/R implant, on the NRT threshold (TNRT) profile across the array. CI24M/R measurements were simulated by ignoring CI24RE measurements with response amplitudes below 50 uV. Comparisons of the estimated TNRTs from the CI24RE measurements and the CI24M/R simulations suggest that, apart from a constant level difference, the TNRT profiles from the newer implant generally would not have differed very much from those of its predecessor. This view was also reflected by principal component analysis (PCA) results which revealed a 'shift' component similar to that reported by Smoorenburg et al (2002). On the whole, there is no indication that current practices of using the TNRT profiles for assisting with speech processor programming need to be revised for the CI24RE implant

Performance of an adaptive beamforming noise reduction scheme for hearing aid applications. II. Experimental verification of the predictions

A method to predict the amount of noise reduction which can be achieved using a two-microphone adaptive beamforming noise reduction system for hearing aids [J. Acoust. Soc. Am. 109, 1123 (2001)] is verified experimentally. 34 experiments are performed in real environments and 58 in simulated environments and the results are compared to the predictions. In all experiments, one noise source and one target signal source are present. Starting from a setting in a moderately reverberant room (reverberation time 0.42 s, volume 34 m3, distance between listener and either sound source 1 m, length of the adaptive filter 25 ms), eight different parameters of the acoustical environment and three different design parameters of the adaptive beamformer were systematically varied. For those experiments, in which the direct-to-reverberant ratios of the noise signal is +3 dB or less, the difference between the predicted and the measured improvement in signal-to-noise ratio (SNR) is -0.21+/-0.59 dB for real environments and -0.25+/-0.51 dB for simulated environments (average +/- standard deviation). At higher direct-to-reverberant ratios, SNR improvement is systematically underestimated by up to 5.34 dB. The parameters with the greatest influence on the performance of the adaptive beamformer have been found to be the direct-to-reverberant ratio of the noise source, the reverberation time of the acoustic environment, and the length of the adaptive filter

Performance of an adaptive beamforming noise reduction scheme for hearing aid applications. I. Prediction of the signal-to-noise-ratio improvement

Adaptive beamformers have been proposed as noise reduction schemes for conventional hearing aids and cochlear implants. A method to predict the amount of noise reduction that can be achieved by a two-microphone adaptive beamformer is presented. The prediction is based on a model of the acoustic environment in which the presence of one acoustic target-signal source and one acoustic noise source in a reverberant enclosure is assumed. The acoustic field is sampled using two omnidirectional microphones mounted close to the ears of a user. The model takes eleven different parameters into account, including reverberation time and size of the room, directionality of the acoustic sources, and design parameters of the beamformer itself, including length of the adaptive filter and delay in the target signal path. An approximation to predict the achievable signal-to-noise improvement based on the model is presented. Potential applications as well as limitations of the proposed prediction method are discussed and a FORTRAN subroutine to predict the achievable signal-to-noise improvement is provided. Experimental verification of the predictions is provided in a companion paper [J. Acoust. Soc. Am. 109, 1134 (2001)]

Neural adaptation and the ECAP response threshold: a pilot study

The electrically evoked compound action potential (ECAP) amplitude resulting from a train of pulses of finite duration (100 ms or 200 ms) was found to vary inversely to the stimulation rate (pulse rate), corroborating well with neural adaptation results from a previous study (Dillier et al., 2005). Amplitude growth functions based on these adapted responses yield thresholds (TNRT) that increase with increasing pulse rate, contrary to behavioural thresholds, which are known to vary inversely with the stimulation rate. Adaptation effects are therefore a confounding factor that must be accounted for when attempting to compare behavioural and objective measures

RFcap: A software analysis tool for multichannel cochlear implant signals

Being able to display and analyse the output of a speech processor that encodes the parameters of complex stimuli to be presented by a cochlear implant (CI) is useful for software and hardware development as well asfor diagnostic purposes. This firstly requires appropriate hardware that is able to receive and decode theradio frequency (RF)-coded signals, and then processing the decoded data using suitable software. The PCI-IF6 clinical hardware for the Nucleus CI system, together with the Nucleus Implant Communicator and Nucleus Matlab Toolbox research software libraries, provide the necessary functionality. RFcap is a standalone Matlab application that encapsulates the relevant functions to capture, display, and analyse the RF-coded signals intended for the Nucleus CI24M/R, CI24RE, and CI500 multichannel CIs

Optimierung von 'sound quality' und Musikhören mit Cochlea-Implantaten

Cochlea-Implantate (CI) wurden in den letzten 20 Jahren bei vollständig und hochgradig tauben Patienten sehr erfolgreich zur Wiederherstellung des Sprachverständnisses eingesetzt, trotz Einschränkungen bezüglich Bandbreite, Klangdynamik sowie der zeitlichen und spektralen Auflösung im Vergleich zu Normalhörenden. Die meisten CI-Träger beurteilen jedoch die Klangqualität beim Musikhören als unbefriedigend und stark verbesserungswürdig. Unter der Annahme, dass eine Verbesserung der Klangqualität auch das Musikhören verbessert, wird hier überblicksmässig dargestellt, wie dieses Ziel mit der heutigen Technik zu erreichen ist und wo deren Grenzen liegen